Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was always defined)

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 914 matching lines @@
         return -1;
     }
 
     // --- Add modules to process thread (for periodic schedulation)
 
     _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get());
 
     // --- ACM initialization
 
     if ((audio_coding_->InitializeReceiver() == -1)
-#ifdef WEBRTC_CODEC_AVT
         // out-of-band Dtmf tones are played out by default
-        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)
-#endif
-       )
-    {
+        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)) {
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
             "Channel::Init() unable to initialize the ACM - 1");
         return -1;
     }
 
     // --- RTP/RTCP module initialization
 
     // Ensure that RTCP is enabled by default for the created channel.
     // Note that, the module will keep generating RTCP until it is explicitly
@@ skipping 3201 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc